Skip to content
Snippets Groups Projects
Select Git revision
  • dev_fxs_no_wb
  • lk_channel_joined_bridge_indication
  • 14600-wenpeng
  • asterisk_rdkb
  • fix_transport_fallback_at_dns_failure
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-test-250121
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_re_register_authorization_header
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • 22.0.0-pre1
  • 21.4.2
  • 20.9.2
  • 18.24.2
  • certified-20.7-cert2
  • certified-18.9-cert11
  • 21.4.1
  • 20.9.1
  • 18.24.1
  • 21.4.0
  • 20.9.0
  • 18.24.0
  • certified-20.7-cert1
  • certified-18.9-cert10
  • 21.4.0-rc1
  • 20.9.0-rc1
  • 18.24.0-rc1
  • 21.3.1
  • 20.8.1
  • 18.23.1
40 results
Search by author
  • Any Author
  • Adam Porteous Adam Porteous adam.porteous
  • Alex Hopkins Alex Hopkins 59hopper
  • Amin Ben Romdhane Amin Ben Romdhane amin.benromdhane
  • Andreas Gnau Andreas Gnau AGnau
  • Andrej Kucharik Andrej Kucharik andrej.kucharik
  • Andrii Pypa Andrii Pypa andrii.pypa
  • Anjan Chanda Anjan Chanda anjancx
  • Balalakshmi Arunachalam Rajendran Balalakshmi Arunachalam Rajendran arbala
  • Benjamin Larsson Benjamin Larsson benjamin
  • Charles Foster Charles Foster charles.foster_bt.com
  • Dariusz Iwanoczko Dariusz Iwanoczko dariusz.iwanoczko
  • Divya Chandrasekaran Divya Chandrasekaran divya.c
  • Dominic Lake Dominic Lake dominic.lake
  • Elena Vengerova Elena Vengerova helen
  • Erik Karlsson Erik Karlsson erik.karlsson
  • Filip Matusiak Filip Matusiak filmat
  • Filip Strelec Filip Strelec filip.strelec
  • Fredrik Östling Fredrik Östling fredrik.ostling
  • Geoff Elliott Geoff Elliott gdelliott
  • George Yang George Yang gyang
20 authors
  1. Apr 11, 2011
  3. Apr 07, 2011
  5. Apr 05, 2011
  7. Apr 01, 2011
    • Richard Mudgett's avatar
      Merged revisions 312461 via svnmerge from · 75594e6e
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
  
  CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
  
  The CallCompletionRequest()/CallCompletionCancel() dialplan applications
  exit nonzero on normal failure conditions.  The nonzero exit causes the
  dialplan to hangup immediately.  The dialplan author has no opportunity to
  report success/failure to the user.
  
  * Made always return zero so the dialplan can continue.
  
  * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
  CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.  Also
  documented the values set.
  
  * Reduced the warning about no core instance in CallCompletionCancel() to
  a debug message.  It is a normal event and should not be output at the
  WARNING level.
  
  (closes issue #18763)
  Reported by: p_lindheimer
  Patches:
        ccss.patch uploaded by p lindheimer (license 558) Modified
  Tested by: p_lindheimer, rmudgett
  
  JIRA SWP-3042
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      75594e6e
    • Jonathan Rose's avatar
      New Feature for chan_dahdi. 4 length pattern matching. · 846cfa0e
      Jonathan Rose authored 
      In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      846cfa0e
    • Tilghman Lesher's avatar
      Merged revisions 312286,312288 via svnmerge from · 3731fd9c
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
  
  Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
................
  r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
  
  Merged revisions 312287 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
    
    Merged revisions 312285 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
      
      Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
      
      (issue #18969)
       Reported by: oej
       Patches: 
             20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3731fd9c
  9. Mar 17, 2011
  11. Mar 16, 2011
  13. Mar 15, 2011
  15. Mar 14, 2011
  17. Mar 11, 2011
  19. Mar 10, 2011
  21. Mar 07, 2011
  23. Mar 05, 2011
  25. Mar 02, 2011
  27. Mar 01, 2011
  29. Feb 28, 2011
  31. Feb 24, 2011
  33. Feb 23, 2011
  35. Feb 22, 2011
    • David Vossel's avatar
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd... · d760e81f
      David Vossel authored 
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d760e81f
    • Andrew Latham's avatar
      Use ast_debug for console logging · 736133f8
      Andrew Latham authored 
      Guessed the log levels based on info that level 3
is the soft roof.  Can we create a page / document
to define the levels?



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      736133f8
  37. Feb 21, 2011
  39. Feb 19, 2011
  41. Feb 15, 2011
    • Richard Mudgett's avatar
      Merged revisions 307879 via svnmerge from · b2ef13cb
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
  
  No response sent for SIP CC subscribe/resubscribe request.
  
  Asterisk does not send a response if we try to subscribe for call
  completion after we have received a 180 Ringing.  You can only subscribe
  for call completion when the call has been cleared.
  
  When we receive the 180 Ringing, for this call, its call-completion state
  is 'CC_AVAILABLE'.  If we then send a subscribe message to Asterisk, it
  trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
  Because this is an invalid state change, it just ignores the message.  The
  only state Asterisk will accept our subscribe message is in the
  'CC_CALLER_OFFERED' state.
  
  Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
  the call by sending a CANCEL.
  
  Asterisk should always send a response.  Even if its a negative one.
  
  
  The fix is to allow for the CCSS core to notify a CC agent that a failure
  has occurred when CC is requested.  The "ack" callback is replaced with a
  "respond" callback.  The "respond" callback has a parameter indicating
  either a successful response or a specific type of failure that may need
  to be communicated to the requester.
  
  (closes issue #18336)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson, rmudgett
  
  JIRA SWP-2633
  
  (closes issue #18337)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson
  
  JIRA SWP-2634
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b2ef13cb
  43. Feb 10, 2011
  45. Feb 09, 2011
  47. Feb 08, 2011